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How outbound calls work with BYOT

Outbound calls from your SIP trunk are placed via the standard Bolna call API. Bolna looks up the from_number, resolves the associated trunk and gateway, and places the call via Asterisk’s PJSIP channel driver using your trunk’s credentials.

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Prerequisites

Before making outbound calls, ensure:
  1. You have created a SIP trunk on Bolna
  2. You have added your phone numbers to the trunk
  3. is_active is true on the trunk
  4. You have a Bolna agent created

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Step 1. Configure the agent’s telephony provider

Set telephony_provider to "sip-trunk" in your agent’s configuration. This tells Bolna to route outbound calls through your SIP trunk and use ulaw audio encoding. 
curl -X PATCH https://api.bolna.ai/v2/agent/{agent_id} \
  -H "Authorization: Bearer <token>" \
  -H "Content-Type: application/json" \
  -d '{
    "agent_config": {
      "telephony_provider": "sip-trunk"
    }
  }'

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Step 2. Place an outbound call

Use the standard call API, specifying the from_number as a DID registered on your trunk: 
curl -X POST https://api.bolna.ai/call \
  -H "Authorization: Bearer <token>" \
  -H "Content-Type: application/json" \
  -d '{
    "agent_id": "agt-xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx",
    "recipient": {
      "phone_number": "+918800001234",
      "name": "Rahul Sharma"
    },
    "from_number": "+919876543210"
  }'
The from_number must be a phone number you have registered on your SIP trunk. Bolna will use the trunk’s gateway and credentials to place the call.

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Troubleshooting outbound calls

If outbound calls are failing or not being placed:
  1. Confirm is_active is true on the trunk
  2. Confirm the from_number used in the call request is registered on the trunk
  3. Confirm the gateway address and credentials are correct — check your provider portal for authentication errors
  4. For ip-based trunks, ensure 13.200.45.61 is in your provider’s IP whitelist
  5. Confirm outbound_leading_plus_enabled matches what your carrier expects (some carriers reject +, others require it)

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No audio / call connects but no voice

SRTP is not supported. If your SIP trunk has SRTP enabled, the media negotiation will fail. Disable SRTP on your trunk and ensure media is negotiated as plain RTP.

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Audio quality issues / one-way audio

  • Ensure rtp_symmetric and force_rport are both true (the defaults)
  • Confirm your SIP provider’s RTP IP ranges are reachable from 13.200.45.61
  • Verify allow includes ulaw
  • Avoid enabling direct_media unless explicitly advised by Bolna support

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Next steps